| 00:10.39 | *** join/#asterisk pancho_jay (~pancho@181.95.214.109) |
| 00:18.44 | pancho_jay | Hello all! |
| 00:18.51 | pancho_jay | I have problems with asterisk11 and odbc |
| 00:18.58 | pancho_jay | I am trying to use realtime configuration for sip peers |
| 00:19.06 | pancho_jay | but asterisk shows me a lot of messages like these: |
| 00:19.12 | pancho_jay | [Mar 5 20:06:31] WARNING[19260]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000: ERROR: column "regseconds" is of type bigint but expression is of type character at character 149; |
| 00:19.21 | pancho_jay | I don't know why Asterisk is trying to save regseconds field as string (instead of integer) |
| 00:19.36 | pancho_jay | I am using PostgreSQL 9.5 and Ubuntu 14.04 64bits |
| 00:35.44 | *** join/#asterisk Garibaldo (~smuxi@177.158.156.86) |
| 01:00.28 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:23.56 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 01:28.18 | *** join/#asterisk Draecos (~Draecos@203-121-194-218.e-wire.net.au) |
| 01:36.58 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 01:39.27 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 02:31.23 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:36.29 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 02:45.24 | *** join/#asterisk jdub_35 (~jonathan@173.218.138.135) |
| 02:45.46 | jdub_35 | hello... is anyone there? |
| 02:51.01 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 03:00.31 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 03:05.45 | *** join/#asterisk Oatmeal (~Suzeanne@50.232.212.36) |
| 03:07.51 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 03:08.56 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 03:18.32 | jdub_35 | anyone there? |
| 03:59.15 | *** join/#asterisk Oatmeal (~Suzeanne@166.175.56.231) |
| 04:14.34 | *** join/#asterisk kunwon1 (~kunwon1@unaffiliated/kunwon1) |
| 04:16.12 | kunwon1 | hi, using asterisk 13.5.0 - with res_hep and res_hep_rtcp - and using ngrep to monitor outgoing traffic. If I unload res_hep_rtcp, there is zero traffic to my sip capture server. if i leave it loaded, i get rtcp HEP packets but nothing else (no registration, call info, etc) - any tips on how to fix? |
| 05:24.00 | drmessano | What is the power supply voltage on a Polycom phone? Like a 330/331 |
| 05:24.04 | drmessano | 24v? |
| 05:26.40 | kunwon1 | drmessano: yes |
| 05:28.49 | drmessano | Thanks.. guess I need to work out the current draw now |
| 05:30.40 | kunwon1 | drmessano: the factory power supply supplies 0.5A |
| 05:30.51 | drmessano | Awesome.. thank you! |
| 05:31.39 | drmessano | I wonder if you can run 24v to them PoE |
| 05:31.50 | drmessano | Im guessing not |
| 05:32.06 | kunwon1 | they run with any modern PoE switch |
| 05:32.54 | drmessano | I know that much |
| 05:33.13 | drmessano | Looks like they dont work with 24v on the ethernet cable |
| 05:33.23 | drmessano | SO I will need a splitter |
| 05:51.36 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 05:58.35 | *** join/#asterisk Oatmeal (~Suzeanne@50.232.212.36) |
| 06:09.21 | *** join/#asterisk Draecos (~Draecos@203-121-194-218.e-wire.net.au) |
| 06:53.56 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 07:12.13 | *** join/#asterisk doome_ (~doome@94-21-148-183.pool.digikabel.hu) |
| 07:15.03 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net) |
| 07:21.03 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:51.41 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 08:02.11 | *** join/#asterisk SaxonManangan (~sapsun@122.49.220.186) |
| 08:05.05 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 08:32.31 | *** join/#asterisk XATRIX (~XATRIX@178.212.64.96) |
| 08:32.43 | *** join/#asterisk areski (~areski@80.174.128.60.dyn.user.ono.com) |
| 08:33.06 | XATRIX | Hi, why when i change my SIP port 5060->6060 my SIP client stops to make\receive calls ? |
| 08:33.29 | XATRIX | I can register on PBX but, i can 't make calls. debug info says : NAT type: Symmetric |
| 08:33.51 | XATRIX | But if i switch back to 5050 ,everything goes ok |
| 08:55.29 | *** join/#asterisk hypknight (~Atlantis@unaffiliated/hypknight) |
| 09:03.04 | drmessano | why are you changing it? |
| 09:03.06 | drmessano | nm |
| 09:30.33 | snadge | 5060 is just too mainstream ;) |
| 09:32.55 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-lkddvwjsdslndual) |
| 09:44.04 | drmessano | Yeah |
| 09:44.16 | drmessano | Gotta keep out the evil hackers |
| 09:53.37 | *** join/#asterisk znf (~ibm86@unu.card-sharing.eu) |
| 09:55.23 | hypknight|Away | is now away - Reason : Auto-Away (Away from Keyboard for 60 minutes) |
| 10:10.54 | wdoekes | hypknight|Away: you're wasting a lot of peoples time by getting attention for that auto-away message in multiple channels |
| 10:11.02 | wdoekes | don't do that |
| 10:11.32 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:42.20 | *** join/#asterisk Oatmeal (~Suzeanne@50.232.212.36) |
| 10:45.37 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 10:46.30 | *** join/#asterisk O47m341 (~Suzeanne@50.232.212.36) |
| 11:12.07 | *** join/#asterisk markit (~marco@88-149-177-66.v4.ngi.it) |
| 11:28.00 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 11:36.00 | *** join/#asterisk O47m341 (~Suzeanne@50.232.212.36) |
| 11:49.20 | *** join/#asterisk areski (~areski@80.174.128.60.dyn.user.ono.com) |
| 12:20.29 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:00.26 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 13:27.45 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 13:44.39 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-ezboggpplghvxnze) |
| 13:53.21 | *** join/#asterisk Draecos (~Draecos@203-121-194-218.e-wire.net.au) |
| 13:59.59 | *** join/#asterisk Andymeows (Andymeows@unaffiliated/andymeows) |
| 14:13.45 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-ypfzqdimmmgmkakk) |
| 14:18.07 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:fde2:8c9a:a1b1:8270) |
| 14:30.01 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:39.11 | *** join/#asterisk pancho_jay (~pancho@181.95.214.109) |
| 14:39.18 | pancho_jay | Hello all! |
| 14:39.55 | pancho_jay | I am trying to setup Asterisk with realtime support for SIP peers, but it isn't working |
| 14:40.10 | pancho_jay | Asterisk shows me a lot of warning messages |
| 14:44.31 | pancho_jay | WARNING[25800]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000: ERROR: column "regseconds" is of type bigint but expression is of type character at character 143; |
| 14:45.09 | pancho_jay | I don't know why Asterisk is trying to save regseconds field as a string. Regseconds should be an integer |
| 15:03.59 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 15:07.57 | *** join/#asterisk areski (~areski@80.174.128.60.dyn.user.ono.com) |
| 15:39.56 | *** join/#asterisk Andymeows (Andymeows@unaffiliated/andymeows) |
| 15:41.42 | *** join/#asterisk KValchev (~kvalchev@cable-84-43-131-215.mnet.bg) |
| 15:59.00 | voip | thank you so much |
| 16:13.25 | *** join/#asterisk protem (~protem@unaffiliated/protem) |
| 16:32.51 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 17:45.16 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 18:13.17 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 18:13.46 | *** join/#asterisk kline (~kline@nds/committee/kline) |
| 18:14.24 | kline | is csipsimple on android known to not work with asterisk, or is the debian package for asterisk just junk? |
| 18:14.27 | *** join/#asterisk tzafrir (~tzafrir@37.142.35.116) |
| 18:14.57 | kline | im following the tutorial on the wiki with the debian packages, and i can make sip extensions and dial in, but all my calls are silent |
| 18:18.35 | kline | I'm pretty sure the calling itself is working - if I change the Wait() parameter, I see matching behaviour in the call |
| 18:20.13 | WIMPy | Any NAT involved? |
| 18:20.17 | WIMPy | ~sipnat |
| 18:20.17 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 18:22.13 | kline | shouldnt be, we're on the same lan. asterisk is running on a vm but the interfaces are bridged rather than natted |
| 18:22.35 | kline | I'll have a read in any case |
| 18:22.44 | WIMPy | Maybe the client is by default configured for NAT support. |
| 18:23.06 | kline | the AO link btw is 404 |
| 18:23.20 | WIMPy | Thw what? |
| 18:23.37 | kline | http://www.aocomputing.net/?p=3 |
| 18:23.55 | kline | from the ~sipnat snippet |
| 18:24.15 | WIMPy | Oh, still :-( |
| 18:36.42 | kline | ok, so im still not 100% convinced theres a nat involved, but csipsimple says that the local nat type is "port restricted" |
| 18:37.15 | WIMPy | Well, it thinks there is. |
| 18:37.53 | WIMPy | That's enough to cause trouble. Turn it off or configure Asterisk as if the client was behind NAT. |
| 18:37.59 | kline | yeah |
| 18:41.55 | kunwon1 | anyone successfully using res_hep? I have enabled that and res_hep_rtcp and only the latter is sending UDP packets to my capture server, there is nothing being generated by res_hep |
| 18:42.28 | kunwon1 | i updated to 13.7.2 and the problem persists |
| 18:44.46 | kline | WIMPy, ok, thanks, that sorted it at least for hello-world |
| 18:54.21 | kunwon1 | looks like res_hep only works with pjsip, not chan_sip |
| 18:57.08 | WIMPy | Sounds likely. |
| 19:02.36 | *** join/#asterisk youtmon (~yout@c-73-46-212-142.hsd1.fl.comcast.net) |
| 19:40.27 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 20:01.10 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 20:22.12 | *** join/#asterisk kasanop (~kasanop@ivanpoddubny.com) |
| 20:22.49 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 20:30.25 | julius | hey |
| 20:32.54 | julius | by chance i got my hands on something called a "easybox", when creating a SIP account in asterisk is there any way to test that without having to attach the box to the phone line? |
| 20:33.44 | julius | i mean like a with a softphone, call the box and have the call redirected to asterisk |
| 20:33.55 | julius | does that make sense at all? |
| 20:34.14 | WIMPy | Not sure. can you even configure that thing? |
| 20:35.05 | WIMPy | And what kind of Box is it? |
| 21:07.27 | *** join/#asterisk danjenkins (danjenkins@gateway/shell/firrre/x-fjnxpaijwnusfuey) |
| 21:07.56 | *** join/#asterisk areski (~areski@80.174.128.60.dyn.user.ono.com) |
| 21:23.09 | *** join/#asterisk kunwon1 (~kunwon1@unaffiliated/kunwon1) |
| 23:10.22 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |